US6243673B1 - Speech coding apparatus and pitch prediction method of input speech signal - Google Patents

Speech coding apparatus and pitch prediction method of input speech signal Download PDF

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US6243673B1
US6243673B1 US09/153,299 US15329998A US6243673B1 US 6243673 B1 US6243673 B1 US 6243673B1 US 15329998 A US15329998 A US 15329998A US 6243673 B1 US6243673 B1 US 6243673B1
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pitch
convolution calculation
excitation pulse
pulse sequence
nth
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Motoyasu Ohno
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Panasonic System Solutions Japan Co Ltd
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Matsushita Graphic Communication Systems Inc
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/09Long term prediction, i.e. removing periodical redundancies, e.g. by using adaptive codebook or pitch predictor
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L2019/0001Codebooks
    • G10L2019/0011Long term prediction filters, i.e. pitch estimation
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L2019/0001Codebooks
    • G10L2019/0013Codebook search algorithms

Definitions

  • the present invention relates to a speech coding apparatus and a pitch prediction method in speech coding, particularly a speech coding apparatus using a pitch prediction method in which pitch information concerning an input excitation waveform for speech coding is obtained as few computations as possible, and a pitch prediction method of an input speech signal.
  • a speech coding method represented by CELP (Code Excited Linear Prediction) system is performed by modelimg the speech information using a speech waveform and an excitation waveform, and coding the spectrum envelop information corresponding to the speech waveform, and the pitch information corresponding to the excitation waveform separately, both of which are extracted from input speech information divided into frames.
  • CELP Code Excited Linear Prediction
  • the coding according to G.723.1 is carried out based on the principles of linear prediction analysis-by-synthesis to attempt so that a perceptually weighted error signal is minimized.
  • the search of pitch information in this case is performed by using the characteristics that a speech waveform changes periodically in a vowel range corresponding to the vibration of a vocal cord, which is called pitch prediction.
  • FIG. 1 is a block diagram of a pitch prediction section in a conventional speech coding apparatus.
  • An input speech signal is processed to be divided into frames and sub-frames.
  • An excitation pulse sequence X[n] generated in a immediately before sub-frame is input to pitch reproduction processing section 1 , and processed by the pitch emphasis processing for a current target sub-frame.
  • Linear predictive synthesis filter 2 provides at multiplier 3 the system filter processing such as formant processing and harmonic shaping processing to an output speech data Y[n] from pitch reproduction processing section 1 .
  • the coefficient setting of this linear predictive synthesis filter 2 is performed using a linear predictive coefficient A′(z) normalized by the LSP (linear spectrum pair) quantization of a linear predictive coefficient A(z) obtained by linear predictive analyzing a speech input signal y[n], a perceptual weighting coefficient W[z] used in perceptual weighting processing the input speech signal y[n], and a coefficient P(z) signal of harmonic noise filter for waveform arranging a perceptually weighted signal.
  • LSP linear spectrum pair
  • Pitch predictive filter 4 is a filter with five taps for providing in multiplier 5 the filter processing to an output data t′[n] out put from multiplier 3 using a predetermined coefficient. This coefficient setting is performed by reading out a codeword sequentially from adaptive codebook 6 in which a codeword of adaptive vector corresponding to each pitch period is stored. Further when coded speech data are decoded, this pitch predictive filter 4 has the function to generate a pitch period which sounds more natural and similar to a human speech in generating a current excitation pulse sequence from a previous excitation pulse sequence.
  • Further adder 7 outputs an error signal r[n].
  • the error signal r[n] is an error between an output data p[n] from multiplier 5 that is a pitch predictive filtering processed signal, and a pitch residual signal t[n] of a current sub-frame (a residual signal of the formant processing and the harmonic shaping processing).
  • An index in adaptive codebook 6 and a pitch length are obtained as the optimal pitch information so that the error signal r[n] should be minimized by the least squares method.
  • the calculation processing in a pitch prediction method described above is performed in the following way.
  • the excitation pulse sequence X[n] of a certain pitch is sequentially input to a buffer to which 145 samples can be input, then the pitch reproduced excitation sequence Y[n] of 64 samples are obtained according to equations (1) and (2) below, where Lag indicates a pitch period.
  • equations (1) and (2) indicate that a current pitch information (vocal cord vibration) is imitated using a previous excitation pulse sequence.
  • the convolution data (filtered data) t′[n] is obtained by the convolution of this pitch reproduced excitation sequence Y[n] and an output from linear predictive synthesis filter 2 according to equation (3) below.
  • l is a variable of two dimensional matrix, which indicates the processing is repeated five times.
  • the optimal value of convolution data P(n) in pitch predictive filter 4 is obtained using pitch residual signal t(n) so that the error signal r(n) should be minimized.
  • the error signal r(n) shown in equation (6) below should be minimized by searching adaptive codebook data of pitches corresponding to Live filter coefficients of fifth order FIR type pitch predictive filter 4 from codebook 6 .
  • adaptive codebook data of a pitch in other words, the index of adaptive codebook data of a pitch to minimize the error is obtained.
  • Further pitch information that is closed loop pitch information and the index of adaptive code book data of a pitch are obtained by repeating the above operation corresponding to Lag ⁇ 1 up to Lag+1 for the re-search so as to obtain the pitch period information at this time correctly.
  • the further processing is provided to each sub-frame.
  • the pitch search processing is performed according to the range described above, and since one frame is composed of four sub-frames, the same processing is repeated four times in one frame.
  • the present invention is carried out by considering the above subjects. It is an object of the present invention to provide a speech coding apparatus using the pitch prediction method capable of reducing the computations in DSP (CPU) without depending on the k parameter.
  • the convolution processing which requires the plurality of computations corresponding to the number of repeating times set by the k parameter, is completed with only one computation. That allows reducing the computations in a CPU.
  • the present invention is to store in advance a plurality of pitch reproduced excitation pulse sequences, to which the pitch reproduction processing is provided, corresponding to a plurality of pitch searches, and to perform the convolution processing sequentially by reading the pitch reproduced excitation pulse from the memory.
  • the pitch searches are simplified since the second time. And since it is not necessary to repeat the pitch reproduction processing according to the k parameter, it is possible to reduce the calculation amount in a CPU.
  • FIG. 1 is a block diagram of a pitch prediction section of a conventional speech coding apparatus
  • FIG. 2 is an exemplary diagram illustrating the state in generating a pitch reproduced excitation sequence
  • FIG. 3 is a block diagram of a pitch prediction section in a speech coding apparatus in the first embodiment of the present invention
  • FIG. 4A is an exemplary diagram illustrating a memory to store convolution data in a speech coding apparatus in the first embodiment
  • FIG. 4B is an exemplary diagram illustrating the state in shifting convolution data in the memory in a speech coding apparatus in the first embodiment.
  • FIG. 5 is a block diagram of a pitch prediction section in a speech coding apparatus in the second embodiment of the present invention.
  • FIG. 3 is a schematic block diagram of a pitch prediction section in a speech coding apparatus in the first embodiment of the present invention.
  • An excitation pulse sequence X[n] generated in a just-previous sub-frame is input pitch reproduction processing section 1 .
  • Pitch reproduction processing section 1 provides the pitch emphasis processing for a current object sub-frame using the input X[n] based on the pitch length information obtained by the auto-correlation of the input speech waveform.
  • linear predictive synthesis filter 2 provides at multiplier 3 the system filter processing such as formant processing and harmonic shaping processing to an output speech data Y[n] from pitch reproduction processing section 1 .
  • the coefficient setting of this linear predictive synthesis filter 2 is performed using a linear predictive coefficient A′(z) normalized by the LSP quantization, a perceptual weighting coefficient W[z] and a coefficient P(z) signal of harmonic noise filter.
  • Pitch predictive filter 4 is a filter with five taps for providing in multiplier 5 the filter processing to an output data t′[n] in multiplier 3 using a predetermined coefficient. This coefficient setting is performed by reading a codeword sequentially from adaptive codebook 6 in which a codeword of adaptive vector corresponding to each pitch period is stored.
  • Further adder 7 outputs an error signal r[n]
  • the error signal r[n] is an error between an output data p[n] from multiplier 5 that is a pitch predictive filter processed signal, and a pitch residual signal t[n] of the current sub-frame (a residual signal after the formant processing and the harmonic shaping processing).
  • An index in adaptive codebook 6 and a pitch length are obtained as the optimal pitch information so that the error signal r[n] is minimized by the least squares method.
  • pitch deciding section 8 detects the pitch period (Lag) from the input pitch length information, and decides whether or not the value exceeds the predetermined value.
  • pitch period (Lag)
  • one sub-frame is composed of 60 samples
  • one period is more than one sub-frame
  • pitch predictive filter is composed of 5 taps
  • And memory 9 is to store the convolution data of the pitch reproduced excitation data Y[n] and a coefficient I[n] of linear predictive synthesis filter 2 .
  • first convolution data up to fifth convolution data are sequentially stored in memory 9 corresponding to the repeating times of pitch reproduction set by the k parameter and the convolution.
  • an excitation pulse sequence X′[n] is feedback to pitch reproduction processing section 2 , using pitch information acquired at the previous processing.
  • the excitation pulse sequence X′[n] is generated from an error signal between the convolution data of the coefficient of pitch predictive filter 4 using the previous convolution data and pitch residual signal t[n].
  • each convolution data of t′(4)(n) according to equation (3) and equation (5) in the first embodiment is the same as that in a conventional technology.
  • the previous pitch reproduction processing result is used again in the case where pitch period Lag is more than a predetermined value when re-search is performed k times by repeating the convolution processing using linear predictive synthesis filter 2 to improve the reproduction precision of a pitch period. That is attempted to reduce the computations.
  • the second pitch reproduction processing is performed in the order of Lag+1, lag and Lag ⁇ 1 according to equation (10) and equation (11) below.
  • the second and third pitch re-search processing is performed in the same manner.
  • this convolution is performed 5 times according to equation (4) and equation (5).
  • the convolution data are sequentially stored in memory 9 .
  • the previous convolution data stored in memory 9 is used in the convolution processing at this time.
  • the fourth convolution data at the previous time are the fifth convolution data at this time
  • the third convolution data at the previous time are the fourth convolution data at this time
  • the second convolution data at the previous time are third convolution data at this time
  • the first convolution data are newly computed and stored in memory 9 as illustrated in FIG. 4 A.
  • the first convolution data up to the fourth convolution data obtained in the first search processing are each copied and respectively stored in the second search data write area in memory 9 . That allows reducing the computations.
  • the fourth convolution data are stored in a storing area for the fifth convolution data that will be unnecessary, then the third and second data are stored sequentially, and finally the first convolution data are computed to store.
  • the memory areas it is possible to reduce the memory areas.
  • the pitch predictive processing can be always performed with five storing areas for the convolution data, which are at least necessary for the fifth order FIR.
  • a memory controller in memory 9 performs the processing described above, i.e., the write of the convolution data to memory 9 , the shift of the convolution data in memory 9 , and the read of convolution data used in the current pitch search from memory 9 .
  • the memory controller is one of functions of memory 9 .
  • the convolution data obtained as described above are returned to a pitch reproduction processing section as closed loop pitch information to be processed by the pitch reproduction processing, and are processed by the convolution processing with the filter coefficient set for linear predictive synthesis filter 2 .
  • Such processing is repeated corresponding to the number of repeating times set by the k parameter. That permits to improve the precision of the pitch reproduction excitation sequence t′[n] to be inputted to multiplier 5 .

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Analogue/Digital Conversion (AREA)
US09/153,299 1997-09-20 1998-09-15 Speech coding apparatus and pitch prediction method of input speech signal Expired - Fee Related US6243673B1 (en)

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JP27373897A JP3263347B2 (ja) 1997-09-20 1997-09-20 音声符号化装置及び音声符号化におけるピッチ予測方法
JP9-273738 1997-09-20

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Cited By (5)

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US20030093266A1 (en) * 2001-11-13 2003-05-15 Matsushita Electric Industrial Co., Ltd. Speech coding apparatus, speech decoding apparatus and speech coding/decoding method
US20030163317A1 (en) * 2001-01-25 2003-08-28 Tetsujiro Kondo Data processing device
US20040117178A1 (en) * 2001-03-07 2004-06-17 Kazunori Ozawa Sound encoding apparatus and method, and sound decoding apparatus and method
US20100286991A1 (en) * 2008-01-04 2010-11-11 Dolby International Ab Audio encoder and decoder
US20100332954A1 (en) * 2009-06-24 2010-12-30 Lsi Corporation Systems and Methods for Out of Order Y-Sample Memory Management

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Publication number Priority date Publication date Assignee Title
CN116615736A (zh) * 2020-09-18 2023-08-18 维萨国际服务协会 经由光卷积进行的动态图节点嵌入

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US20030163317A1 (en) * 2001-01-25 2003-08-28 Tetsujiro Kondo Data processing device
US7269559B2 (en) * 2001-01-25 2007-09-11 Sony Corporation Speech decoding apparatus and method using prediction and class taps
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US20100286991A1 (en) * 2008-01-04 2010-11-11 Dolby International Ab Audio encoder and decoder
US20100286990A1 (en) * 2008-01-04 2010-11-11 Dolby International Ab Audio encoder and decoder
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US8494863B2 (en) * 2008-01-04 2013-07-23 Dolby Laboratories Licensing Corporation Audio encoder and decoder with long term prediction
US8924201B2 (en) 2008-01-04 2014-12-30 Dolby International Ab Audio encoder and decoder
US8938387B2 (en) 2008-01-04 2015-01-20 Dolby Laboratories Licensing Corporation Audio encoder and decoder
US20100332954A1 (en) * 2009-06-24 2010-12-30 Lsi Corporation Systems and Methods for Out of Order Y-Sample Memory Management
US8352841B2 (en) * 2009-06-24 2013-01-08 Lsi Corporation Systems and methods for out of order Y-sample memory management

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JP3263347B2 (ja) 2002-03-04
JPH1195799A (ja) 1999-04-09
DE69822579D1 (de) 2004-04-29
EP0903729B1 (de) 2004-03-24
DE69822579T2 (de) 2004-08-05
EP0903729A3 (de) 1999-12-29
EP0903729A2 (de) 1999-03-24

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